2009-01-25 18:20:15 +00:00
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/*-
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* Copyright 2008 by Marco Trillo. All rights reserved.
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*
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* Redistribution and use in source and binary forms, with or without
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* modification, are permitted provided that the following conditions
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* are met:
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* 1. Redistributions of source code must retain the above copyright
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* notice, this list of conditions and the following disclaimer.
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* 2. Redistributions in binary form must reproduce the above copyright
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* notice, this list of conditions and the following disclaimer in the
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* documentation and/or other materials provided with the distribution.
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*
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* THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR
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* IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES
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* OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED.
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* IN NO EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT,
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* INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING,
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* BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES;
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* LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED
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* AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY,
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* OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY
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* OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF
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* SUCH DAMAGE.
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*
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* $FreeBSD$
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*/
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/*
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* Apple DAVbus audio controller.
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*/
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#include <sys/param.h>
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#include <sys/systm.h>
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#include <sys/bus.h>
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#include <sys/kernel.h>
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#include <sys/lock.h>
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#include <sys/malloc.h>
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#include <sys/module.h>
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#include <sys/mutex.h>
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#include <sys/rman.h>
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#include <dev/ofw/ofw_bus.h>
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Sound Mega-commit. Expect further cleanup until code freeze.
For a slightly thorough explaination, please refer to
[1] http://people.freebsd.org/~ariff/SOUND_4.TXT.html .
Summary of changes includes:
1 Volume Per-Channel (vpc). Provides private / standalone volume control
unique per-stream pcm channel without touching master volume / pcm.
Applications can directly use SNDCTL_DSP_[GET|SET][PLAY|REC]VOL, or for
backwards compatibility, SOUND_MIXER_PCM through the opened dsp device
instead of /dev/mixer. Special "bypass" mode is enabled through
/dev/mixer which will automatically detect if the adjustment is made
through /dev/mixer and forward its request to this private volume
controller. Changes to this volume object will not interfere with
other channels.
Requirements:
- SNDCTL_DSP_[GET|SET][PLAY|REC]_VOL are newer ioctls (OSSv4) which
require specific application modifications (preferred).
- No modifications required for using bypass mode, so applications
like mplayer or xmms should work out of the box.
Kernel hints:
- hint.pcm.%d.vpc (0 = disable vpc).
Kernel sysctls:
- hw.snd.vpc_mixer_bypass (default: 1). Enable or disable /dev/mixer
bypass mode.
- hw.snd.vpc_autoreset (default: 1). By default, closing/opening
/dev/dsp will reset the volume back to 0 db gain/attenuation.
Setting this to 0 will preserve its settings across device
closing/opening.
- hw.snd.vpc_reset (default: 0). Panic/reset button to reset all
volume settings back to 0 db.
- hw.snd.vpc_0db (default: 45). 0 db relative to linear mixer value.
2 High quality fixed-point Bandlimited SINC sampling rate converter,
based on Julius O'Smith's Digital Audio Resampling -
http://ccrma.stanford.edu/~jos/resample/. It includes a filter design
script written in awk (the clumsiest joke I've ever written)
- 100% 32bit fixed-point, 64bit accumulator.
- Possibly among the fastest (if not fastest) of its kind.
- Resampling quality is tunable, either runtime or during kernel
compilation (FEEDER_RATE_PRESETS).
- Quality can be further customized during kernel compilation by
defining FEEDER_RATE_PRESETS in /etc/make.conf.
Kernel sysctls:
- hw.snd.feeder_rate_quality.
0 - Zero-order Hold (ZOH). Fastest, bad quality.
1 - Linear Interpolation (LINEAR). Slightly slower than ZOH,
better quality but still does not eliminate aliasing.
2 - (and above) - Sinc Interpolation(SINC). Best quality. SINC
quality always start from 2 and above.
Rough quality comparisons:
- http://people.freebsd.org/~ariff/z_comparison/
3 Bit-perfect mode. Bypasses all feeder/dsp effects. Pure sound will be
directly fed into the hardware.
4 Parametric (compile time) Software Equalizer (Bass/Treble mixer). Can
be customized by defining FEEDER_EQ_PRESETS in /etc/make.conf.
5 Transparent/Adaptive Virtual Channel. Now you don't have to disable
vchans in order to make digital format pass through. It also makes
vchans more dynamic by choosing a better format/rate among all the
concurrent streams, which means that dev.pcm.X.play.vchanformat/rate
becomes sort of optional.
6 Exclusive Stream, with special open() mode O_EXCL. This will "mute"
other concurrent vchan streams and only allow a single channel with
O_EXCL set to keep producing sound.
Other Changes:
* most feeder_* stuffs are compilable in userland. Let's not
speculate whether we should go all out for it (save that for
FreeBSD 16.0-RELEASE).
* kobj signature fixups, thanks to Andriy Gapon <avg@freebsd.org>
* pull out channel mixing logic out of vchan.c and create its own
feeder_mixer for world justice.
* various refactoring here and there, for good or bad.
* activation of few more OSSv4 ioctls() (see [1] above).
* opt_snd.h for possible compile time configuration:
(mostly for debugging purposes, don't try these at home)
SND_DEBUG
SND_DIAGNOSTIC
SND_FEEDER_MULTIFORMAT
SND_FEEDER_FULL_MULTIFORMAT
SND_FEEDER_RATE_HP
SND_PCM_64
SND_OLDSTEREO
Manual page updates are on the way.
Tested by: joel, Olivier SMEDTS <olivier at gid0 d org>, too many
unsung / unnamed heroes.
2009-06-07 19:12:08 +00:00
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#ifdef HAVE_KERNEL_OPTION_HEADERS
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#include "opt_snd.h"
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#endif
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2009-01-25 18:20:15 +00:00
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#include <dev/sound/pcm/sound.h>
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Sound Mega-commit. Expect further cleanup until code freeze.
For a slightly thorough explaination, please refer to
[1] http://people.freebsd.org/~ariff/SOUND_4.TXT.html .
Summary of changes includes:
1 Volume Per-Channel (vpc). Provides private / standalone volume control
unique per-stream pcm channel without touching master volume / pcm.
Applications can directly use SNDCTL_DSP_[GET|SET][PLAY|REC]VOL, or for
backwards compatibility, SOUND_MIXER_PCM through the opened dsp device
instead of /dev/mixer. Special "bypass" mode is enabled through
/dev/mixer which will automatically detect if the adjustment is made
through /dev/mixer and forward its request to this private volume
controller. Changes to this volume object will not interfere with
other channels.
Requirements:
- SNDCTL_DSP_[GET|SET][PLAY|REC]_VOL are newer ioctls (OSSv4) which
require specific application modifications (preferred).
- No modifications required for using bypass mode, so applications
like mplayer or xmms should work out of the box.
Kernel hints:
- hint.pcm.%d.vpc (0 = disable vpc).
Kernel sysctls:
- hw.snd.vpc_mixer_bypass (default: 1). Enable or disable /dev/mixer
bypass mode.
- hw.snd.vpc_autoreset (default: 1). By default, closing/opening
/dev/dsp will reset the volume back to 0 db gain/attenuation.
Setting this to 0 will preserve its settings across device
closing/opening.
- hw.snd.vpc_reset (default: 0). Panic/reset button to reset all
volume settings back to 0 db.
- hw.snd.vpc_0db (default: 45). 0 db relative to linear mixer value.
2 High quality fixed-point Bandlimited SINC sampling rate converter,
based on Julius O'Smith's Digital Audio Resampling -
http://ccrma.stanford.edu/~jos/resample/. It includes a filter design
script written in awk (the clumsiest joke I've ever written)
- 100% 32bit fixed-point, 64bit accumulator.
- Possibly among the fastest (if not fastest) of its kind.
- Resampling quality is tunable, either runtime or during kernel
compilation (FEEDER_RATE_PRESETS).
- Quality can be further customized during kernel compilation by
defining FEEDER_RATE_PRESETS in /etc/make.conf.
Kernel sysctls:
- hw.snd.feeder_rate_quality.
0 - Zero-order Hold (ZOH). Fastest, bad quality.
1 - Linear Interpolation (LINEAR). Slightly slower than ZOH,
better quality but still does not eliminate aliasing.
2 - (and above) - Sinc Interpolation(SINC). Best quality. SINC
quality always start from 2 and above.
Rough quality comparisons:
- http://people.freebsd.org/~ariff/z_comparison/
3 Bit-perfect mode. Bypasses all feeder/dsp effects. Pure sound will be
directly fed into the hardware.
4 Parametric (compile time) Software Equalizer (Bass/Treble mixer). Can
be customized by defining FEEDER_EQ_PRESETS in /etc/make.conf.
5 Transparent/Adaptive Virtual Channel. Now you don't have to disable
vchans in order to make digital format pass through. It also makes
vchans more dynamic by choosing a better format/rate among all the
concurrent streams, which means that dev.pcm.X.play.vchanformat/rate
becomes sort of optional.
6 Exclusive Stream, with special open() mode O_EXCL. This will "mute"
other concurrent vchan streams and only allow a single channel with
O_EXCL set to keep producing sound.
Other Changes:
* most feeder_* stuffs are compilable in userland. Let's not
speculate whether we should go all out for it (save that for
FreeBSD 16.0-RELEASE).
* kobj signature fixups, thanks to Andriy Gapon <avg@freebsd.org>
* pull out channel mixing logic out of vchan.c and create its own
feeder_mixer for world justice.
* various refactoring here and there, for good or bad.
* activation of few more OSSv4 ioctls() (see [1] above).
* opt_snd.h for possible compile time configuration:
(mostly for debugging purposes, don't try these at home)
SND_DEBUG
SND_DIAGNOSTIC
SND_FEEDER_MULTIFORMAT
SND_FEEDER_FULL_MULTIFORMAT
SND_FEEDER_RATE_HP
SND_PCM_64
SND_OLDSTEREO
Manual page updates are on the way.
Tested by: joel, Olivier SMEDTS <olivier at gid0 d org>, too many
unsung / unnamed heroes.
2009-06-07 19:12:08 +00:00
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2009-01-25 18:20:15 +00:00
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#include <dev/sound/macio/aoa.h>
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#include <dev/sound/macio/davbusreg.h>
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#include <machine/intr_machdep.h>
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#include <machine/resource.h>
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#include <machine/bus.h>
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#include "mixer_if.h"
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struct davbus_softc {
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struct aoa_softc aoa;
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phandle_t node;
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phandle_t soundnode;
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struct resource *reg;
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struct mtx mutex;
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int device_id;
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u_int output_mask;
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u_int (*read_status)(struct davbus_softc *, u_int);
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void (*set_outputs)(struct davbus_softc *, u_int);
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};
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static int davbus_probe(device_t);
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static int davbus_attach(device_t);
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static void davbus_cint(void *);
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static device_method_t pcm_davbus_methods[] = {
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/* Device interface. */
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DEVMETHOD(device_probe, davbus_probe),
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DEVMETHOD(device_attach, davbus_attach),
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{ 0, 0 }
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};
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static driver_t pcm_davbus_driver = {
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"pcm",
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pcm_davbus_methods,
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2009-01-26 14:43:18 +00:00
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PCM_SOFTC_SIZE
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2009-01-25 18:20:15 +00:00
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};
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DRIVER_MODULE(pcm_davbus, macio, pcm_davbus_driver, pcm_devclass, 0, 0);
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MODULE_DEPEND(pcm_davbus, sound, SOUND_MINVER, SOUND_PREFVER, SOUND_MAXVER);
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/*****************************************************************************
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Probe and attachment routines.
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*****************************************************************************/
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static int
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davbus_probe(device_t self)
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{
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const char *name;
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name = ofw_bus_get_name(self);
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if (!name)
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return (ENXIO);
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if (strcmp(name, "davbus") != 0)
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return (ENXIO);
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device_set_desc(self, "Apple DAVBus Audio Controller");
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return (0);
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}
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/*
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* Burgundy codec control
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*/
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static int burgundy_init(struct snd_mixer *m);
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static int burgundy_uninit(struct snd_mixer *m);
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static int burgundy_reinit(struct snd_mixer *m);
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static void burgundy_write_locked(struct davbus_softc *, u_int, u_int);
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static void burgundy_set_outputs(struct davbus_softc *d, u_int mask);
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static u_int burgundy_read_status(struct davbus_softc *d, u_int status);
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static int burgundy_set(struct snd_mixer *m, unsigned dev, unsigned left,
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unsigned right);
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Sound Mega-commit. Expect further cleanup until code freeze.
For a slightly thorough explaination, please refer to
[1] http://people.freebsd.org/~ariff/SOUND_4.TXT.html .
Summary of changes includes:
1 Volume Per-Channel (vpc). Provides private / standalone volume control
unique per-stream pcm channel without touching master volume / pcm.
Applications can directly use SNDCTL_DSP_[GET|SET][PLAY|REC]VOL, or for
backwards compatibility, SOUND_MIXER_PCM through the opened dsp device
instead of /dev/mixer. Special "bypass" mode is enabled through
/dev/mixer which will automatically detect if the adjustment is made
through /dev/mixer and forward its request to this private volume
controller. Changes to this volume object will not interfere with
other channels.
Requirements:
- SNDCTL_DSP_[GET|SET][PLAY|REC]_VOL are newer ioctls (OSSv4) which
require specific application modifications (preferred).
- No modifications required for using bypass mode, so applications
like mplayer or xmms should work out of the box.
Kernel hints:
- hint.pcm.%d.vpc (0 = disable vpc).
Kernel sysctls:
- hw.snd.vpc_mixer_bypass (default: 1). Enable or disable /dev/mixer
bypass mode.
- hw.snd.vpc_autoreset (default: 1). By default, closing/opening
/dev/dsp will reset the volume back to 0 db gain/attenuation.
Setting this to 0 will preserve its settings across device
closing/opening.
- hw.snd.vpc_reset (default: 0). Panic/reset button to reset all
volume settings back to 0 db.
- hw.snd.vpc_0db (default: 45). 0 db relative to linear mixer value.
2 High quality fixed-point Bandlimited SINC sampling rate converter,
based on Julius O'Smith's Digital Audio Resampling -
http://ccrma.stanford.edu/~jos/resample/. It includes a filter design
script written in awk (the clumsiest joke I've ever written)
- 100% 32bit fixed-point, 64bit accumulator.
- Possibly among the fastest (if not fastest) of its kind.
- Resampling quality is tunable, either runtime or during kernel
compilation (FEEDER_RATE_PRESETS).
- Quality can be further customized during kernel compilation by
defining FEEDER_RATE_PRESETS in /etc/make.conf.
Kernel sysctls:
- hw.snd.feeder_rate_quality.
0 - Zero-order Hold (ZOH). Fastest, bad quality.
1 - Linear Interpolation (LINEAR). Slightly slower than ZOH,
better quality but still does not eliminate aliasing.
2 - (and above) - Sinc Interpolation(SINC). Best quality. SINC
quality always start from 2 and above.
Rough quality comparisons:
- http://people.freebsd.org/~ariff/z_comparison/
3 Bit-perfect mode. Bypasses all feeder/dsp effects. Pure sound will be
directly fed into the hardware.
4 Parametric (compile time) Software Equalizer (Bass/Treble mixer). Can
be customized by defining FEEDER_EQ_PRESETS in /etc/make.conf.
5 Transparent/Adaptive Virtual Channel. Now you don't have to disable
vchans in order to make digital format pass through. It also makes
vchans more dynamic by choosing a better format/rate among all the
concurrent streams, which means that dev.pcm.X.play.vchanformat/rate
becomes sort of optional.
6 Exclusive Stream, with special open() mode O_EXCL. This will "mute"
other concurrent vchan streams and only allow a single channel with
O_EXCL set to keep producing sound.
Other Changes:
* most feeder_* stuffs are compilable in userland. Let's not
speculate whether we should go all out for it (save that for
FreeBSD 16.0-RELEASE).
* kobj signature fixups, thanks to Andriy Gapon <avg@freebsd.org>
* pull out channel mixing logic out of vchan.c and create its own
feeder_mixer for world justice.
* various refactoring here and there, for good or bad.
* activation of few more OSSv4 ioctls() (see [1] above).
* opt_snd.h for possible compile time configuration:
(mostly for debugging purposes, don't try these at home)
SND_DEBUG
SND_DIAGNOSTIC
SND_FEEDER_MULTIFORMAT
SND_FEEDER_FULL_MULTIFORMAT
SND_FEEDER_RATE_HP
SND_PCM_64
SND_OLDSTEREO
Manual page updates are on the way.
Tested by: joel, Olivier SMEDTS <olivier at gid0 d org>, too many
unsung / unnamed heroes.
2009-06-07 19:12:08 +00:00
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static u_int32_t burgundy_setrecsrc(struct snd_mixer *m, u_int32_t src);
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2009-01-25 18:20:15 +00:00
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static kobj_method_t burgundy_mixer_methods[] = {
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KOBJMETHOD(mixer_init, burgundy_init),
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KOBJMETHOD(mixer_uninit, burgundy_uninit),
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KOBJMETHOD(mixer_reinit, burgundy_reinit),
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KOBJMETHOD(mixer_set, burgundy_set),
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KOBJMETHOD(mixer_setrecsrc, burgundy_setrecsrc),
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Sound Mega-commit. Expect further cleanup until code freeze.
For a slightly thorough explaination, please refer to
[1] http://people.freebsd.org/~ariff/SOUND_4.TXT.html .
Summary of changes includes:
1 Volume Per-Channel (vpc). Provides private / standalone volume control
unique per-stream pcm channel without touching master volume / pcm.
Applications can directly use SNDCTL_DSP_[GET|SET][PLAY|REC]VOL, or for
backwards compatibility, SOUND_MIXER_PCM through the opened dsp device
instead of /dev/mixer. Special "bypass" mode is enabled through
/dev/mixer which will automatically detect if the adjustment is made
through /dev/mixer and forward its request to this private volume
controller. Changes to this volume object will not interfere with
other channels.
Requirements:
- SNDCTL_DSP_[GET|SET][PLAY|REC]_VOL are newer ioctls (OSSv4) which
require specific application modifications (preferred).
- No modifications required for using bypass mode, so applications
like mplayer or xmms should work out of the box.
Kernel hints:
- hint.pcm.%d.vpc (0 = disable vpc).
Kernel sysctls:
- hw.snd.vpc_mixer_bypass (default: 1). Enable or disable /dev/mixer
bypass mode.
- hw.snd.vpc_autoreset (default: 1). By default, closing/opening
/dev/dsp will reset the volume back to 0 db gain/attenuation.
Setting this to 0 will preserve its settings across device
closing/opening.
- hw.snd.vpc_reset (default: 0). Panic/reset button to reset all
volume settings back to 0 db.
- hw.snd.vpc_0db (default: 45). 0 db relative to linear mixer value.
2 High quality fixed-point Bandlimited SINC sampling rate converter,
based on Julius O'Smith's Digital Audio Resampling -
http://ccrma.stanford.edu/~jos/resample/. It includes a filter design
script written in awk (the clumsiest joke I've ever written)
- 100% 32bit fixed-point, 64bit accumulator.
- Possibly among the fastest (if not fastest) of its kind.
- Resampling quality is tunable, either runtime or during kernel
compilation (FEEDER_RATE_PRESETS).
- Quality can be further customized during kernel compilation by
defining FEEDER_RATE_PRESETS in /etc/make.conf.
Kernel sysctls:
- hw.snd.feeder_rate_quality.
0 - Zero-order Hold (ZOH). Fastest, bad quality.
1 - Linear Interpolation (LINEAR). Slightly slower than ZOH,
better quality but still does not eliminate aliasing.
2 - (and above) - Sinc Interpolation(SINC). Best quality. SINC
quality always start from 2 and above.
Rough quality comparisons:
- http://people.freebsd.org/~ariff/z_comparison/
3 Bit-perfect mode. Bypasses all feeder/dsp effects. Pure sound will be
directly fed into the hardware.
4 Parametric (compile time) Software Equalizer (Bass/Treble mixer). Can
be customized by defining FEEDER_EQ_PRESETS in /etc/make.conf.
5 Transparent/Adaptive Virtual Channel. Now you don't have to disable
vchans in order to make digital format pass through. It also makes
vchans more dynamic by choosing a better format/rate among all the
concurrent streams, which means that dev.pcm.X.play.vchanformat/rate
becomes sort of optional.
6 Exclusive Stream, with special open() mode O_EXCL. This will "mute"
other concurrent vchan streams and only allow a single channel with
O_EXCL set to keep producing sound.
Other Changes:
* most feeder_* stuffs are compilable in userland. Let's not
speculate whether we should go all out for it (save that for
FreeBSD 16.0-RELEASE).
* kobj signature fixups, thanks to Andriy Gapon <avg@freebsd.org>
* pull out channel mixing logic out of vchan.c and create its own
feeder_mixer for world justice.
* various refactoring here and there, for good or bad.
* activation of few more OSSv4 ioctls() (see [1] above).
* opt_snd.h for possible compile time configuration:
(mostly for debugging purposes, don't try these at home)
SND_DEBUG
SND_DIAGNOSTIC
SND_FEEDER_MULTIFORMAT
SND_FEEDER_FULL_MULTIFORMAT
SND_FEEDER_RATE_HP
SND_PCM_64
SND_OLDSTEREO
Manual page updates are on the way.
Tested by: joel, Olivier SMEDTS <olivier at gid0 d org>, too many
unsung / unnamed heroes.
2009-06-07 19:12:08 +00:00
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KOBJMETHOD_END
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2009-01-25 18:20:15 +00:00
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};
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MIXER_DECLARE(burgundy_mixer);
|
|
|
|
|
|
|
|
static int
|
|
|
|
burgundy_init(struct snd_mixer *m)
|
|
|
|
{
|
|
|
|
struct davbus_softc *d;
|
|
|
|
|
|
|
|
d = mix_getdevinfo(m);
|
|
|
|
|
|
|
|
d->read_status = burgundy_read_status;
|
|
|
|
d->set_outputs = burgundy_set_outputs;
|
|
|
|
|
|
|
|
/*
|
|
|
|
* We configure the Burgundy codec as follows:
|
|
|
|
*
|
|
|
|
* o Input subframe 0 is connected to input digital
|
|
|
|
* stream A (ISA).
|
|
|
|
* o Stream A (ISA) is mixed in mixer 2 (MIX2).
|
|
|
|
* o Output of mixer 2 (MIX2) is routed to output sources
|
|
|
|
* OS0 and OS1 which can be converted to analog.
|
|
|
|
*
|
|
|
|
*/
|
|
|
|
mtx_lock(&d->mutex);
|
|
|
|
|
|
|
|
burgundy_write_locked(d, 0x16700, 0x40);
|
|
|
|
|
|
|
|
burgundy_write_locked(d, BURGUNDY_MIX0_REG, 0);
|
|
|
|
burgundy_write_locked(d, BURGUNDY_MIX1_REG, 0);
|
|
|
|
burgundy_write_locked(d, BURGUNDY_MIX2_REG, BURGUNDY_MIX_ISA);
|
|
|
|
burgundy_write_locked(d, BURGUNDY_MIX3_REG, 0);
|
|
|
|
|
|
|
|
burgundy_write_locked(d, BURGUNDY_OS_REG, BURGUNDY_OS0_MIX2 |
|
|
|
|
BURGUNDY_OS1_MIX2);
|
|
|
|
|
|
|
|
burgundy_write_locked(d, BURGUNDY_SDIN_REG, BURGUNDY_ISA_SF0);
|
|
|
|
|
|
|
|
/* Set several digital scalers to unity gain. */
|
|
|
|
burgundy_write_locked(d, BURGUNDY_MXS2L_REG, BURGUNDY_MXS_UNITY);
|
|
|
|
burgundy_write_locked(d, BURGUNDY_MXS2R_REG, BURGUNDY_MXS_UNITY);
|
|
|
|
burgundy_write_locked(d, BURGUNDY_OSS0L_REG, BURGUNDY_OSS_UNITY);
|
|
|
|
burgundy_write_locked(d, BURGUNDY_OSS0R_REG, BURGUNDY_OSS_UNITY);
|
|
|
|
burgundy_write_locked(d, BURGUNDY_OSS1L_REG, BURGUNDY_OSS_UNITY);
|
|
|
|
burgundy_write_locked(d, BURGUNDY_OSS1R_REG, BURGUNDY_OSS_UNITY);
|
|
|
|
burgundy_write_locked(d, BURGUNDY_ISSAL_REG, BURGUNDY_ISS_UNITY);
|
|
|
|
burgundy_write_locked(d, BURGUNDY_ISSAR_REG, BURGUNDY_ISS_UNITY);
|
|
|
|
|
|
|
|
burgundy_set_outputs(d, burgundy_read_status(d,
|
|
|
|
bus_read_4(d->reg, DAVBUS_CODEC_STATUS)));
|
|
|
|
|
|
|
|
mtx_unlock(&d->mutex);
|
|
|
|
|
|
|
|
mix_setdevs(m, SOUND_MASK_VOLUME);
|
|
|
|
|
|
|
|
return (0);
|
|
|
|
}
|
|
|
|
|
|
|
|
static int
|
|
|
|
burgundy_uninit(struct snd_mixer *m)
|
|
|
|
{
|
|
|
|
return (0);
|
|
|
|
}
|
|
|
|
|
|
|
|
static int
|
|
|
|
burgundy_reinit(struct snd_mixer *m)
|
|
|
|
{
|
|
|
|
return (0);
|
|
|
|
}
|
|
|
|
|
|
|
|
static void
|
|
|
|
burgundy_write_locked(struct davbus_softc *d, u_int reg, u_int val)
|
|
|
|
{
|
|
|
|
u_int size, addr, offset, data, i;
|
|
|
|
|
|
|
|
size = (reg & 0x00FF0000) >> 16;
|
|
|
|
addr = (reg & 0x0000FF00) >> 8;
|
|
|
|
offset = reg & 0xFF;
|
|
|
|
|
|
|
|
for (i = offset; i < offset + size; ++i) {
|
|
|
|
data = BURGUNDY_CTRL_WRITE | (addr << 12) |
|
|
|
|
((size + offset - 1) << 10) | (i << 8) | (val & 0xFF);
|
|
|
|
if (i == offset)
|
|
|
|
data |= BURGUNDY_CTRL_RESET;
|
|
|
|
|
|
|
|
bus_write_4(d->reg, DAVBUS_CODEC_CTRL, data);
|
|
|
|
|
|
|
|
while (bus_read_4(d->reg, DAVBUS_CODEC_CTRL) &
|
|
|
|
DAVBUS_CODEC_BUSY)
|
|
|
|
DELAY(1);
|
|
|
|
|
|
|
|
val >>= 8; /* next byte. */
|
|
|
|
}
|
|
|
|
}
|
|
|
|
|
|
|
|
/* Must be called with d->mutex held. */
|
|
|
|
static void
|
|
|
|
burgundy_set_outputs(struct davbus_softc *d, u_int mask)
|
|
|
|
{
|
|
|
|
u_int x = 0;
|
|
|
|
|
|
|
|
if (mask == d->output_mask)
|
|
|
|
return;
|
|
|
|
|
|
|
|
/*
|
|
|
|
* Bordeaux card wirings:
|
|
|
|
* Port 15: RCA out
|
|
|
|
* Port 16: Minijack out
|
|
|
|
* Port 17: Internal speaker
|
|
|
|
*
|
|
|
|
* B&W G3 wirings:
|
|
|
|
* Port 14: Minijack out
|
|
|
|
* Port 17: Internal speaker
|
|
|
|
*/
|
|
|
|
|
|
|
|
DPRINTF(("Enabled outputs:"));
|
|
|
|
if (mask & (1 << 0)) {
|
|
|
|
DPRINTF((" SPEAKER"));
|
|
|
|
x |= BURGUNDY_P17M_EN;
|
|
|
|
}
|
|
|
|
if (mask & (1 << 1)) {
|
|
|
|
DPRINTF((" HEADPHONES"));
|
|
|
|
x |= BURGUNDY_P14L_EN | BURGUNDY_P14R_EN;
|
|
|
|
}
|
|
|
|
DPRINTF(("\n"));
|
|
|
|
|
|
|
|
burgundy_write_locked(d, BURGUNDY_MUTE_REG, x);
|
|
|
|
d->output_mask = mask;
|
|
|
|
}
|
|
|
|
|
|
|
|
static u_int
|
|
|
|
burgundy_read_status(struct davbus_softc *d, u_int status)
|
|
|
|
{
|
|
|
|
if (status & 0x4)
|
|
|
|
return (1 << 1);
|
|
|
|
else
|
|
|
|
return (1 << 0);
|
|
|
|
}
|
|
|
|
|
|
|
|
static int
|
|
|
|
burgundy_set(struct snd_mixer *m, unsigned dev, unsigned left, unsigned right)
|
|
|
|
{
|
|
|
|
struct davbus_softc *d;
|
|
|
|
int lval, rval;
|
|
|
|
|
|
|
|
lval = ((100 - left) * 15 / 100) & 0xf;
|
|
|
|
rval = ((100 - right) * 15 / 100) & 0xf;
|
|
|
|
DPRINTF(("volume %d %d\n", lval, rval));
|
|
|
|
|
|
|
|
d = mix_getdevinfo(m);
|
|
|
|
|
|
|
|
switch (dev) {
|
|
|
|
case SOUND_MIXER_VOLUME:
|
|
|
|
mtx_lock(&d->mutex);
|
|
|
|
|
|
|
|
burgundy_write_locked(d, BURGUNDY_OL13_REG, lval);
|
|
|
|
burgundy_write_locked(d, BURGUNDY_OL14_REG, (rval << 4) | lval);
|
|
|
|
burgundy_write_locked(d, BURGUNDY_OL15_REG, (rval << 4) | lval);
|
|
|
|
burgundy_write_locked(d, BURGUNDY_OL16_REG, (rval << 4) | lval);
|
|
|
|
burgundy_write_locked(d, BURGUNDY_OL17_REG, lval);
|
|
|
|
|
|
|
|
mtx_unlock(&d->mutex);
|
|
|
|
|
|
|
|
return (left | (right << 8));
|
|
|
|
}
|
|
|
|
|
|
|
|
return (0);
|
|
|
|
}
|
|
|
|
|
Sound Mega-commit. Expect further cleanup until code freeze.
For a slightly thorough explaination, please refer to
[1] http://people.freebsd.org/~ariff/SOUND_4.TXT.html .
Summary of changes includes:
1 Volume Per-Channel (vpc). Provides private / standalone volume control
unique per-stream pcm channel without touching master volume / pcm.
Applications can directly use SNDCTL_DSP_[GET|SET][PLAY|REC]VOL, or for
backwards compatibility, SOUND_MIXER_PCM through the opened dsp device
instead of /dev/mixer. Special "bypass" mode is enabled through
/dev/mixer which will automatically detect if the adjustment is made
through /dev/mixer and forward its request to this private volume
controller. Changes to this volume object will not interfere with
other channels.
Requirements:
- SNDCTL_DSP_[GET|SET][PLAY|REC]_VOL are newer ioctls (OSSv4) which
require specific application modifications (preferred).
- No modifications required for using bypass mode, so applications
like mplayer or xmms should work out of the box.
Kernel hints:
- hint.pcm.%d.vpc (0 = disable vpc).
Kernel sysctls:
- hw.snd.vpc_mixer_bypass (default: 1). Enable or disable /dev/mixer
bypass mode.
- hw.snd.vpc_autoreset (default: 1). By default, closing/opening
/dev/dsp will reset the volume back to 0 db gain/attenuation.
Setting this to 0 will preserve its settings across device
closing/opening.
- hw.snd.vpc_reset (default: 0). Panic/reset button to reset all
volume settings back to 0 db.
- hw.snd.vpc_0db (default: 45). 0 db relative to linear mixer value.
2 High quality fixed-point Bandlimited SINC sampling rate converter,
based on Julius O'Smith's Digital Audio Resampling -
http://ccrma.stanford.edu/~jos/resample/. It includes a filter design
script written in awk (the clumsiest joke I've ever written)
- 100% 32bit fixed-point, 64bit accumulator.
- Possibly among the fastest (if not fastest) of its kind.
- Resampling quality is tunable, either runtime or during kernel
compilation (FEEDER_RATE_PRESETS).
- Quality can be further customized during kernel compilation by
defining FEEDER_RATE_PRESETS in /etc/make.conf.
Kernel sysctls:
- hw.snd.feeder_rate_quality.
0 - Zero-order Hold (ZOH). Fastest, bad quality.
1 - Linear Interpolation (LINEAR). Slightly slower than ZOH,
better quality but still does not eliminate aliasing.
2 - (and above) - Sinc Interpolation(SINC). Best quality. SINC
quality always start from 2 and above.
Rough quality comparisons:
- http://people.freebsd.org/~ariff/z_comparison/
3 Bit-perfect mode. Bypasses all feeder/dsp effects. Pure sound will be
directly fed into the hardware.
4 Parametric (compile time) Software Equalizer (Bass/Treble mixer). Can
be customized by defining FEEDER_EQ_PRESETS in /etc/make.conf.
5 Transparent/Adaptive Virtual Channel. Now you don't have to disable
vchans in order to make digital format pass through. It also makes
vchans more dynamic by choosing a better format/rate among all the
concurrent streams, which means that dev.pcm.X.play.vchanformat/rate
becomes sort of optional.
6 Exclusive Stream, with special open() mode O_EXCL. This will "mute"
other concurrent vchan streams and only allow a single channel with
O_EXCL set to keep producing sound.
Other Changes:
* most feeder_* stuffs are compilable in userland. Let's not
speculate whether we should go all out for it (save that for
FreeBSD 16.0-RELEASE).
* kobj signature fixups, thanks to Andriy Gapon <avg@freebsd.org>
* pull out channel mixing logic out of vchan.c and create its own
feeder_mixer for world justice.
* various refactoring here and there, for good or bad.
* activation of few more OSSv4 ioctls() (see [1] above).
* opt_snd.h for possible compile time configuration:
(mostly for debugging purposes, don't try these at home)
SND_DEBUG
SND_DIAGNOSTIC
SND_FEEDER_MULTIFORMAT
SND_FEEDER_FULL_MULTIFORMAT
SND_FEEDER_RATE_HP
SND_PCM_64
SND_OLDSTEREO
Manual page updates are on the way.
Tested by: joel, Olivier SMEDTS <olivier at gid0 d org>, too many
unsung / unnamed heroes.
2009-06-07 19:12:08 +00:00
|
|
|
static u_int32_t
|
2009-01-25 18:20:15 +00:00
|
|
|
burgundy_setrecsrc(struct snd_mixer *m, u_int32_t src)
|
|
|
|
{
|
|
|
|
return (0);
|
|
|
|
}
|
|
|
|
|
|
|
|
/*
|
|
|
|
* Screamer Codec Control
|
|
|
|
*/
|
|
|
|
|
|
|
|
static int screamer_init(struct snd_mixer *m);
|
|
|
|
static int screamer_uninit(struct snd_mixer *m);
|
|
|
|
static int screamer_reinit(struct snd_mixer *m);
|
|
|
|
static void screamer_write_locked(struct davbus_softc *, u_int, u_int);
|
|
|
|
static void screamer_set_outputs(struct davbus_softc *d, u_int mask);
|
|
|
|
static u_int screamer_read_status(struct davbus_softc *d, u_int status);
|
|
|
|
static int screamer_set(struct snd_mixer *m, unsigned dev, unsigned left,
|
|
|
|
unsigned right);
|
Sound Mega-commit. Expect further cleanup until code freeze.
For a slightly thorough explaination, please refer to
[1] http://people.freebsd.org/~ariff/SOUND_4.TXT.html .
Summary of changes includes:
1 Volume Per-Channel (vpc). Provides private / standalone volume control
unique per-stream pcm channel without touching master volume / pcm.
Applications can directly use SNDCTL_DSP_[GET|SET][PLAY|REC]VOL, or for
backwards compatibility, SOUND_MIXER_PCM through the opened dsp device
instead of /dev/mixer. Special "bypass" mode is enabled through
/dev/mixer which will automatically detect if the adjustment is made
through /dev/mixer and forward its request to this private volume
controller. Changes to this volume object will not interfere with
other channels.
Requirements:
- SNDCTL_DSP_[GET|SET][PLAY|REC]_VOL are newer ioctls (OSSv4) which
require specific application modifications (preferred).
- No modifications required for using bypass mode, so applications
like mplayer or xmms should work out of the box.
Kernel hints:
- hint.pcm.%d.vpc (0 = disable vpc).
Kernel sysctls:
- hw.snd.vpc_mixer_bypass (default: 1). Enable or disable /dev/mixer
bypass mode.
- hw.snd.vpc_autoreset (default: 1). By default, closing/opening
/dev/dsp will reset the volume back to 0 db gain/attenuation.
Setting this to 0 will preserve its settings across device
closing/opening.
- hw.snd.vpc_reset (default: 0). Panic/reset button to reset all
volume settings back to 0 db.
- hw.snd.vpc_0db (default: 45). 0 db relative to linear mixer value.
2 High quality fixed-point Bandlimited SINC sampling rate converter,
based on Julius O'Smith's Digital Audio Resampling -
http://ccrma.stanford.edu/~jos/resample/. It includes a filter design
script written in awk (the clumsiest joke I've ever written)
- 100% 32bit fixed-point, 64bit accumulator.
- Possibly among the fastest (if not fastest) of its kind.
- Resampling quality is tunable, either runtime or during kernel
compilation (FEEDER_RATE_PRESETS).
- Quality can be further customized during kernel compilation by
defining FEEDER_RATE_PRESETS in /etc/make.conf.
Kernel sysctls:
- hw.snd.feeder_rate_quality.
0 - Zero-order Hold (ZOH). Fastest, bad quality.
1 - Linear Interpolation (LINEAR). Slightly slower than ZOH,
better quality but still does not eliminate aliasing.
2 - (and above) - Sinc Interpolation(SINC). Best quality. SINC
quality always start from 2 and above.
Rough quality comparisons:
- http://people.freebsd.org/~ariff/z_comparison/
3 Bit-perfect mode. Bypasses all feeder/dsp effects. Pure sound will be
directly fed into the hardware.
4 Parametric (compile time) Software Equalizer (Bass/Treble mixer). Can
be customized by defining FEEDER_EQ_PRESETS in /etc/make.conf.
5 Transparent/Adaptive Virtual Channel. Now you don't have to disable
vchans in order to make digital format pass through. It also makes
vchans more dynamic by choosing a better format/rate among all the
concurrent streams, which means that dev.pcm.X.play.vchanformat/rate
becomes sort of optional.
6 Exclusive Stream, with special open() mode O_EXCL. This will "mute"
other concurrent vchan streams and only allow a single channel with
O_EXCL set to keep producing sound.
Other Changes:
* most feeder_* stuffs are compilable in userland. Let's not
speculate whether we should go all out for it (save that for
FreeBSD 16.0-RELEASE).
* kobj signature fixups, thanks to Andriy Gapon <avg@freebsd.org>
* pull out channel mixing logic out of vchan.c and create its own
feeder_mixer for world justice.
* various refactoring here and there, for good or bad.
* activation of few more OSSv4 ioctls() (see [1] above).
* opt_snd.h for possible compile time configuration:
(mostly for debugging purposes, don't try these at home)
SND_DEBUG
SND_DIAGNOSTIC
SND_FEEDER_MULTIFORMAT
SND_FEEDER_FULL_MULTIFORMAT
SND_FEEDER_RATE_HP
SND_PCM_64
SND_OLDSTEREO
Manual page updates are on the way.
Tested by: joel, Olivier SMEDTS <olivier at gid0 d org>, too many
unsung / unnamed heroes.
2009-06-07 19:12:08 +00:00
|
|
|
static u_int32_t screamer_setrecsrc(struct snd_mixer *m, u_int32_t src);
|
2009-01-25 18:20:15 +00:00
|
|
|
|
|
|
|
static kobj_method_t screamer_mixer_methods[] = {
|
|
|
|
KOBJMETHOD(mixer_init, screamer_init),
|
|
|
|
KOBJMETHOD(mixer_uninit, screamer_uninit),
|
|
|
|
KOBJMETHOD(mixer_reinit, screamer_reinit),
|
|
|
|
KOBJMETHOD(mixer_set, screamer_set),
|
|
|
|
KOBJMETHOD(mixer_setrecsrc, screamer_setrecsrc),
|
Sound Mega-commit. Expect further cleanup until code freeze.
For a slightly thorough explaination, please refer to
[1] http://people.freebsd.org/~ariff/SOUND_4.TXT.html .
Summary of changes includes:
1 Volume Per-Channel (vpc). Provides private / standalone volume control
unique per-stream pcm channel without touching master volume / pcm.
Applications can directly use SNDCTL_DSP_[GET|SET][PLAY|REC]VOL, or for
backwards compatibility, SOUND_MIXER_PCM through the opened dsp device
instead of /dev/mixer. Special "bypass" mode is enabled through
/dev/mixer which will automatically detect if the adjustment is made
through /dev/mixer and forward its request to this private volume
controller. Changes to this volume object will not interfere with
other channels.
Requirements:
- SNDCTL_DSP_[GET|SET][PLAY|REC]_VOL are newer ioctls (OSSv4) which
require specific application modifications (preferred).
- No modifications required for using bypass mode, so applications
like mplayer or xmms should work out of the box.
Kernel hints:
- hint.pcm.%d.vpc (0 = disable vpc).
Kernel sysctls:
- hw.snd.vpc_mixer_bypass (default: 1). Enable or disable /dev/mixer
bypass mode.
- hw.snd.vpc_autoreset (default: 1). By default, closing/opening
/dev/dsp will reset the volume back to 0 db gain/attenuation.
Setting this to 0 will preserve its settings across device
closing/opening.
- hw.snd.vpc_reset (default: 0). Panic/reset button to reset all
volume settings back to 0 db.
- hw.snd.vpc_0db (default: 45). 0 db relative to linear mixer value.
2 High quality fixed-point Bandlimited SINC sampling rate converter,
based on Julius O'Smith's Digital Audio Resampling -
http://ccrma.stanford.edu/~jos/resample/. It includes a filter design
script written in awk (the clumsiest joke I've ever written)
- 100% 32bit fixed-point, 64bit accumulator.
- Possibly among the fastest (if not fastest) of its kind.
- Resampling quality is tunable, either runtime or during kernel
compilation (FEEDER_RATE_PRESETS).
- Quality can be further customized during kernel compilation by
defining FEEDER_RATE_PRESETS in /etc/make.conf.
Kernel sysctls:
- hw.snd.feeder_rate_quality.
0 - Zero-order Hold (ZOH). Fastest, bad quality.
1 - Linear Interpolation (LINEAR). Slightly slower than ZOH,
better quality but still does not eliminate aliasing.
2 - (and above) - Sinc Interpolation(SINC). Best quality. SINC
quality always start from 2 and above.
Rough quality comparisons:
- http://people.freebsd.org/~ariff/z_comparison/
3 Bit-perfect mode. Bypasses all feeder/dsp effects. Pure sound will be
directly fed into the hardware.
4 Parametric (compile time) Software Equalizer (Bass/Treble mixer). Can
be customized by defining FEEDER_EQ_PRESETS in /etc/make.conf.
5 Transparent/Adaptive Virtual Channel. Now you don't have to disable
vchans in order to make digital format pass through. It also makes
vchans more dynamic by choosing a better format/rate among all the
concurrent streams, which means that dev.pcm.X.play.vchanformat/rate
becomes sort of optional.
6 Exclusive Stream, with special open() mode O_EXCL. This will "mute"
other concurrent vchan streams and only allow a single channel with
O_EXCL set to keep producing sound.
Other Changes:
* most feeder_* stuffs are compilable in userland. Let's not
speculate whether we should go all out for it (save that for
FreeBSD 16.0-RELEASE).
* kobj signature fixups, thanks to Andriy Gapon <avg@freebsd.org>
* pull out channel mixing logic out of vchan.c and create its own
feeder_mixer for world justice.
* various refactoring here and there, for good or bad.
* activation of few more OSSv4 ioctls() (see [1] above).
* opt_snd.h for possible compile time configuration:
(mostly for debugging purposes, don't try these at home)
SND_DEBUG
SND_DIAGNOSTIC
SND_FEEDER_MULTIFORMAT
SND_FEEDER_FULL_MULTIFORMAT
SND_FEEDER_RATE_HP
SND_PCM_64
SND_OLDSTEREO
Manual page updates are on the way.
Tested by: joel, Olivier SMEDTS <olivier at gid0 d org>, too many
unsung / unnamed heroes.
2009-06-07 19:12:08 +00:00
|
|
|
KOBJMETHOD_END
|
2009-01-25 18:20:15 +00:00
|
|
|
};
|
|
|
|
|
|
|
|
MIXER_DECLARE(screamer_mixer);
|
|
|
|
|
|
|
|
static int
|
|
|
|
screamer_init(struct snd_mixer *m)
|
|
|
|
{
|
|
|
|
struct davbus_softc *d;
|
|
|
|
|
|
|
|
d = mix_getdevinfo(m);
|
|
|
|
|
|
|
|
d->read_status = screamer_read_status;
|
|
|
|
d->set_outputs = screamer_set_outputs;
|
|
|
|
|
|
|
|
mtx_lock(&d->mutex);
|
|
|
|
|
|
|
|
screamer_write_locked(d, SCREAMER_CODEC_ADDR0, SCREAMER_INPUT_CD |
|
|
|
|
SCREAMER_DEFAULT_CD_GAIN);
|
|
|
|
|
|
|
|
screamer_set_outputs(d, screamer_read_status(d,
|
|
|
|
bus_read_4(d->reg, DAVBUS_CODEC_STATUS)));
|
|
|
|
|
|
|
|
screamer_write_locked(d, SCREAMER_CODEC_ADDR2, 0);
|
|
|
|
screamer_write_locked(d, SCREAMER_CODEC_ADDR4, 0);
|
|
|
|
screamer_write_locked(d, SCREAMER_CODEC_ADDR5, 0);
|
|
|
|
screamer_write_locked(d, SCREAMER_CODEC_ADDR6, 0);
|
|
|
|
|
|
|
|
mtx_unlock(&d->mutex);
|
|
|
|
|
|
|
|
mix_setdevs(m, SOUND_MASK_VOLUME);
|
|
|
|
|
|
|
|
return (0);
|
|
|
|
}
|
|
|
|
|
|
|
|
static int
|
|
|
|
screamer_uninit(struct snd_mixer *m)
|
|
|
|
{
|
|
|
|
return (0);
|
|
|
|
}
|
|
|
|
|
|
|
|
static int
|
|
|
|
screamer_reinit(struct snd_mixer *m)
|
|
|
|
{
|
|
|
|
return (0);
|
|
|
|
}
|
|
|
|
|
|
|
|
|
|
|
|
static void
|
|
|
|
screamer_write_locked(struct davbus_softc *d, u_int reg, u_int val)
|
|
|
|
{
|
|
|
|
u_int x;
|
|
|
|
|
|
|
|
KASSERT(val == (val & 0xfff), ("bad val"));
|
|
|
|
|
|
|
|
while (bus_read_4(d->reg, DAVBUS_CODEC_CTRL) & DAVBUS_CODEC_BUSY)
|
|
|
|
DELAY(100);
|
|
|
|
|
|
|
|
x = reg;
|
|
|
|
x |= SCREAMER_CODEC_EMSEL0;
|
|
|
|
x |= val;
|
|
|
|
bus_write_4(d->reg, DAVBUS_CODEC_CTRL, x);
|
|
|
|
|
|
|
|
while (bus_read_4(d->reg, DAVBUS_CODEC_CTRL) & DAVBUS_CODEC_BUSY)
|
|
|
|
DELAY(100);
|
|
|
|
}
|
|
|
|
|
|
|
|
/* Must be called with d->mutex held. */
|
|
|
|
static void
|
|
|
|
screamer_set_outputs(struct davbus_softc *d, u_int mask)
|
|
|
|
{
|
|
|
|
u_int x;
|
|
|
|
|
|
|
|
if (mask == d->output_mask) {
|
|
|
|
return;
|
|
|
|
}
|
|
|
|
|
|
|
|
x = SCREAMER_MUTE_SPEAKER | SCREAMER_MUTE_HEADPHONES;
|
|
|
|
|
|
|
|
DPRINTF(("Enabled outputs: "));
|
|
|
|
|
|
|
|
if (mask & (1 << 0)) {
|
|
|
|
DPRINTF(("SPEAKER "));
|
|
|
|
x &= ~SCREAMER_MUTE_SPEAKER;
|
|
|
|
}
|
|
|
|
if (mask & (1 << 1)) {
|
|
|
|
DPRINTF(("HEADPHONES "));
|
|
|
|
x &= ~SCREAMER_MUTE_HEADPHONES;
|
|
|
|
}
|
|
|
|
|
|
|
|
DPRINTF(("\n"));
|
|
|
|
|
|
|
|
if (d->device_id == 5 || d->device_id == 11) {
|
|
|
|
DPRINTF(("Enabling programmable output.\n"));
|
|
|
|
x |= SCREAMER_PROG_OUTPUT0;
|
|
|
|
}
|
|
|
|
if (d->device_id == 8 || d->device_id == 11) {
|
|
|
|
x &= ~SCREAMER_MUTE_SPEAKER;
|
|
|
|
|
|
|
|
if (mask & (1 << 0))
|
|
|
|
x |= SCREAMER_PROG_OUTPUT1; /* enable speaker. */
|
|
|
|
}
|
|
|
|
|
|
|
|
screamer_write_locked(d, SCREAMER_CODEC_ADDR1, x);
|
|
|
|
d->output_mask = mask;
|
|
|
|
}
|
|
|
|
|
|
|
|
static u_int
|
|
|
|
screamer_read_status(struct davbus_softc *d, u_int status)
|
|
|
|
{
|
|
|
|
int headphones;
|
|
|
|
|
|
|
|
switch (d->device_id) {
|
|
|
|
case 5: /* Sawtooth */
|
|
|
|
headphones = (status & 0x4);
|
|
|
|
break;
|
|
|
|
|
|
|
|
case 8:
|
|
|
|
case 11: /* iMac DV */
|
|
|
|
/* The iMac DV has 2 headphone outputs. */
|
|
|
|
headphones = (status & 0x7);
|
|
|
|
break;
|
|
|
|
|
|
|
|
default:
|
|
|
|
headphones = (status & 0x8);
|
|
|
|
}
|
|
|
|
|
|
|
|
if (headphones)
|
|
|
|
return (1 << 1);
|
|
|
|
else
|
|
|
|
return (1 << 0);
|
|
|
|
}
|
|
|
|
|
|
|
|
static int
|
|
|
|
screamer_set(struct snd_mixer *m, unsigned dev, unsigned left, unsigned right)
|
|
|
|
{
|
|
|
|
struct davbus_softc *d;
|
|
|
|
int lval, rval;
|
|
|
|
|
|
|
|
lval = ((100 - left) * 15 / 100) & 0xf;
|
|
|
|
rval = ((100 - right) * 15 / 100) & 0xf;
|
|
|
|
DPRINTF(("volume %d %d\n", lval, rval));
|
|
|
|
|
|
|
|
d = mix_getdevinfo(m);
|
|
|
|
|
|
|
|
switch (dev) {
|
|
|
|
case SOUND_MIXER_VOLUME:
|
|
|
|
mtx_lock(&d->mutex);
|
|
|
|
screamer_write_locked(d, SCREAMER_CODEC_ADDR2, (lval << 6) |
|
|
|
|
rval);
|
|
|
|
screamer_write_locked(d, SCREAMER_CODEC_ADDR4, (lval << 6) |
|
|
|
|
rval);
|
|
|
|
mtx_unlock(&d->mutex);
|
|
|
|
|
|
|
|
return (left | (right << 8));
|
|
|
|
}
|
|
|
|
|
|
|
|
return (0);
|
|
|
|
}
|
|
|
|
|
Sound Mega-commit. Expect further cleanup until code freeze.
For a slightly thorough explaination, please refer to
[1] http://people.freebsd.org/~ariff/SOUND_4.TXT.html .
Summary of changes includes:
1 Volume Per-Channel (vpc). Provides private / standalone volume control
unique per-stream pcm channel without touching master volume / pcm.
Applications can directly use SNDCTL_DSP_[GET|SET][PLAY|REC]VOL, or for
backwards compatibility, SOUND_MIXER_PCM through the opened dsp device
instead of /dev/mixer. Special "bypass" mode is enabled through
/dev/mixer which will automatically detect if the adjustment is made
through /dev/mixer and forward its request to this private volume
controller. Changes to this volume object will not interfere with
other channels.
Requirements:
- SNDCTL_DSP_[GET|SET][PLAY|REC]_VOL are newer ioctls (OSSv4) which
require specific application modifications (preferred).
- No modifications required for using bypass mode, so applications
like mplayer or xmms should work out of the box.
Kernel hints:
- hint.pcm.%d.vpc (0 = disable vpc).
Kernel sysctls:
- hw.snd.vpc_mixer_bypass (default: 1). Enable or disable /dev/mixer
bypass mode.
- hw.snd.vpc_autoreset (default: 1). By default, closing/opening
/dev/dsp will reset the volume back to 0 db gain/attenuation.
Setting this to 0 will preserve its settings across device
closing/opening.
- hw.snd.vpc_reset (default: 0). Panic/reset button to reset all
volume settings back to 0 db.
- hw.snd.vpc_0db (default: 45). 0 db relative to linear mixer value.
2 High quality fixed-point Bandlimited SINC sampling rate converter,
based on Julius O'Smith's Digital Audio Resampling -
http://ccrma.stanford.edu/~jos/resample/. It includes a filter design
script written in awk (the clumsiest joke I've ever written)
- 100% 32bit fixed-point, 64bit accumulator.
- Possibly among the fastest (if not fastest) of its kind.
- Resampling quality is tunable, either runtime or during kernel
compilation (FEEDER_RATE_PRESETS).
- Quality can be further customized during kernel compilation by
defining FEEDER_RATE_PRESETS in /etc/make.conf.
Kernel sysctls:
- hw.snd.feeder_rate_quality.
0 - Zero-order Hold (ZOH). Fastest, bad quality.
1 - Linear Interpolation (LINEAR). Slightly slower than ZOH,
better quality but still does not eliminate aliasing.
2 - (and above) - Sinc Interpolation(SINC). Best quality. SINC
quality always start from 2 and above.
Rough quality comparisons:
- http://people.freebsd.org/~ariff/z_comparison/
3 Bit-perfect mode. Bypasses all feeder/dsp effects. Pure sound will be
directly fed into the hardware.
4 Parametric (compile time) Software Equalizer (Bass/Treble mixer). Can
be customized by defining FEEDER_EQ_PRESETS in /etc/make.conf.
5 Transparent/Adaptive Virtual Channel. Now you don't have to disable
vchans in order to make digital format pass through. It also makes
vchans more dynamic by choosing a better format/rate among all the
concurrent streams, which means that dev.pcm.X.play.vchanformat/rate
becomes sort of optional.
6 Exclusive Stream, with special open() mode O_EXCL. This will "mute"
other concurrent vchan streams and only allow a single channel with
O_EXCL set to keep producing sound.
Other Changes:
* most feeder_* stuffs are compilable in userland. Let's not
speculate whether we should go all out for it (save that for
FreeBSD 16.0-RELEASE).
* kobj signature fixups, thanks to Andriy Gapon <avg@freebsd.org>
* pull out channel mixing logic out of vchan.c and create its own
feeder_mixer for world justice.
* various refactoring here and there, for good or bad.
* activation of few more OSSv4 ioctls() (see [1] above).
* opt_snd.h for possible compile time configuration:
(mostly for debugging purposes, don't try these at home)
SND_DEBUG
SND_DIAGNOSTIC
SND_FEEDER_MULTIFORMAT
SND_FEEDER_FULL_MULTIFORMAT
SND_FEEDER_RATE_HP
SND_PCM_64
SND_OLDSTEREO
Manual page updates are on the way.
Tested by: joel, Olivier SMEDTS <olivier at gid0 d org>, too many
unsung / unnamed heroes.
2009-06-07 19:12:08 +00:00
|
|
|
static u_int32_t
|
2009-01-25 18:20:15 +00:00
|
|
|
screamer_setrecsrc(struct snd_mixer *m, u_int32_t src)
|
|
|
|
{
|
|
|
|
return (0);
|
|
|
|
}
|
|
|
|
|
|
|
|
static int
|
|
|
|
davbus_attach(device_t self)
|
|
|
|
{
|
2009-01-26 14:43:18 +00:00
|
|
|
struct davbus_softc *sc;
|
2009-01-25 18:20:15 +00:00
|
|
|
struct resource *dbdma_irq, *cintr;
|
|
|
|
void *cookie;
|
|
|
|
char compat[64];
|
|
|
|
int rid, oirq, err;
|
|
|
|
|
2009-01-26 14:43:18 +00:00
|
|
|
sc = malloc(sizeof(*sc), M_DEVBUF, M_WAITOK | M_ZERO);
|
|
|
|
|
2009-02-07 01:15:13 +00:00
|
|
|
sc->aoa.sc_dev = self;
|
2009-01-25 18:20:15 +00:00
|
|
|
sc->node = ofw_bus_get_node(self);
|
|
|
|
sc->soundnode = OF_child(sc->node);
|
|
|
|
|
|
|
|
/* Map the controller register space. */
|
|
|
|
rid = 0;
|
|
|
|
sc->reg = bus_alloc_resource_any(self, SYS_RES_MEMORY, &rid, RF_ACTIVE);
|
|
|
|
if (sc->reg == NULL)
|
|
|
|
return (ENXIO);
|
|
|
|
|
|
|
|
/* Map the DBDMA channel register space. */
|
|
|
|
rid = 1;
|
|
|
|
sc->aoa.sc_odma = bus_alloc_resource_any(self, SYS_RES_MEMORY,
|
|
|
|
&rid, RF_ACTIVE);
|
|
|
|
if (sc->aoa.sc_odma == NULL)
|
|
|
|
return (ENXIO);
|
|
|
|
|
|
|
|
/* Establish the DBDMA channel edge-triggered interrupt. */
|
|
|
|
rid = 1;
|
|
|
|
dbdma_irq = bus_alloc_resource_any(self, SYS_RES_IRQ,
|
|
|
|
&rid, RF_SHAREABLE | RF_ACTIVE);
|
|
|
|
if (dbdma_irq == NULL)
|
|
|
|
return (ENXIO);
|
|
|
|
|
|
|
|
oirq = rman_get_start(dbdma_irq);
|
|
|
|
|
|
|
|
DPRINTF(("interrupting at irq %d\n", oirq));
|
|
|
|
|
|
|
|
err = powerpc_config_intr(oirq, INTR_TRIGGER_EDGE, INTR_POLARITY_LOW);
|
|
|
|
if (err != 0)
|
|
|
|
return (err);
|
|
|
|
|
2009-02-07 01:15:13 +00:00
|
|
|
snd_setup_intr(self, dbdma_irq, INTR_MPSAFE, aoa_interrupt,
|
|
|
|
sc, &cookie);
|
2009-01-25 18:20:15 +00:00
|
|
|
|
|
|
|
/* Now initialize the controller. */
|
|
|
|
|
|
|
|
bzero(compat, sizeof(compat));
|
|
|
|
OF_getprop(sc->soundnode, "compatible", compat, sizeof(compat));
|
|
|
|
OF_getprop(sc->soundnode, "device-id", &sc->device_id, sizeof(u_int));
|
|
|
|
|
|
|
|
mtx_init(&sc->mutex, "DAVbus", NULL, MTX_DEF);
|
|
|
|
|
|
|
|
device_printf(self, "codec: <%s>\n", compat);
|
|
|
|
|
|
|
|
/* Setup the control interrupt. */
|
|
|
|
rid = 0;
|
|
|
|
cintr = bus_alloc_resource_any(self, SYS_RES_IRQ,
|
|
|
|
&rid, RF_SHAREABLE | RF_ACTIVE);
|
|
|
|
if (cintr != NULL)
|
|
|
|
bus_setup_intr(self, cintr, INTR_TYPE_MISC | INTR_MPSAFE,
|
|
|
|
NULL, davbus_cint, sc, &cookie);
|
|
|
|
|
|
|
|
/* Initialize controller registers. */
|
|
|
|
bus_write_4(sc->reg, DAVBUS_SOUND_CTRL, DAVBUS_INPUT_SUBFRAME0 |
|
|
|
|
DAVBUS_OUTPUT_SUBFRAME0 | DAVBUS_RATE_44100 | DAVBUS_INTR_PORTCHG);
|
|
|
|
|
|
|
|
/* Attach DBDMA engine and PCM layer */
|
2009-02-07 01:15:13 +00:00
|
|
|
err = aoa_attach(sc);
|
2009-01-25 18:20:15 +00:00
|
|
|
if (err)
|
|
|
|
return (err);
|
|
|
|
|
|
|
|
/* Install codec module */
|
|
|
|
if (strcmp(compat, "screamer") == 0)
|
|
|
|
mixer_init(self, &screamer_mixer_class, sc);
|
|
|
|
else if (strcmp(compat, "burgundy") == 0)
|
|
|
|
mixer_init(self, &burgundy_mixer_class, sc);
|
|
|
|
|
|
|
|
return (0);
|
|
|
|
}
|
|
|
|
|
|
|
|
static void
|
|
|
|
davbus_cint(void *ptr)
|
|
|
|
{
|
|
|
|
struct davbus_softc *d = ptr;
|
|
|
|
u_int reg, status, mask;
|
|
|
|
|
|
|
|
mtx_lock(&d->mutex);
|
|
|
|
|
|
|
|
reg = bus_read_4(d->reg, DAVBUS_SOUND_CTRL);
|
|
|
|
if (reg & DAVBUS_PORTCHG) {
|
|
|
|
|
|
|
|
status = bus_read_4(d->reg, DAVBUS_CODEC_STATUS);
|
|
|
|
|
|
|
|
if (d->read_status && d->set_outputs) {
|
|
|
|
|
|
|
|
mask = (*d->read_status)(d, status);
|
|
|
|
(*d->set_outputs)(d, mask);
|
|
|
|
}
|
|
|
|
|
|
|
|
/* Clear the interrupt. */
|
|
|
|
bus_write_4(d->reg, DAVBUS_SOUND_CTRL, reg);
|
|
|
|
}
|
|
|
|
|
|
|
|
mtx_unlock(&d->mutex);
|
|
|
|
}
|
|
|
|
|